On legacy phone systems you can find the following kinds of paging:

• Dial a code to connect to a separate overhead paging and announcement system (like in an airport)
• Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones
• Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones

Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

• New in Asterisk 1.2: The new dialplan command Asterisk cmd Page utilizes MeetMe to page one or more phones.
• New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.

The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.

Advanced Paging / Intercom

There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,… allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

If a phone is in use when a page arrives, some systems can do a “whisper page” so that only the person being paged can hear the page.

SIP phones for the most part don’t support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

• Digital Acoustics Intercoms and VoIP Paging Products
• Aastra/Sayson 480i, 9133i, 480i CT Cordless 5xi phones supported with firmware update March 2008 – Configuration Instructions: Aastra/Sayson AutoAnswer
• Cisco 7940/7960 – see: Cisco 7940-7960 auto-answer config
• Polycom 600 – see: Polycom auto-answer config
• Grandstream GXP2000 and BudgeTone phone supports Auto Answer with firmware 1.0.4.54 or later. GXP-2000 supports true intercom via Call-Info header as of firmware 1.0.1.13.
• Sipura SPA-841 as of firmware version 0.9.5 supports Auto Answer
• Snom Phones see this mailing list posting

Intercom DOES work with the Snom 200 as the mailing list link above shows. Tested on 12/20/04 with firmware 2.04g on Snom 200. One change for that posting is that the variable called in the dialplan must read “_VXML_URL” instead of “VXML_URL”. However, the ‘correct’ way of doing Paging/Intercom is with SipAddHeader. See allpage.agi for example code.

Some analog phones have an Auto Answer function. These phone are often used in door phone systems.

ADSI phones can be configured to Auto Answer if sent the right set of signals.

Some older analog answering machines have a remote intercom function that can be used for overhead paging. Examples: (( Setting up paging using an answering machine))

For overhead paging, you can make an Asterisk Extension go to the sound card, and wire its output to a traditional external paging system. You can also get boxes to interface an phone FXO or FXS port directly to a sound system. Examples:

• Digital Acoustics Website VoIP Paging Products Digital Acoustics (The IP7-ST requires the TalkMaster-SMG software to be used with a SIP PBX)
• Radio Design Labs ST-TC1 Telephone System Coupler. It connects to an FXO port, so more suitable for interface with PBX or other phone system. Also sold as: Smarthome Product
• Bogen TAMB: Connecting TAMB2 tutorial as paging interface with Asterisk PBX.

Another possibility for overhead paging is using hallway speakers that have a direct VOIP connection. Examples:

• Digital Acoustics PoE Paging Speaker for Hallways
• Algo Communication Products Ltd. PoE SIP speakers for voice paging, loud ringing and alert notification
• CyberData PoE VoIP Ceiling Speaker
• Advanced Network Devices PoE IP Speakers

Automated Method

Another way to automate this is with Backticks. Someone has posted a method of using Backticks and shell scripts to dial all phones automatically.

Direct Soundcard Connection

Another method for overhead paging is to solder a cable, with an RCA jack(or whatever you need), directly to the speaker of a phone that provides auto-answer. This cable can be connected directly to your amp or sound system used for paging.

Setting up paging with a sound card

Grandstream Paging

You can use the Grandstream Budgetone phone mentioned above, it even has a round punch-out that can be used to run your cable through. Using the Grandstream as interface to the paging system is a low-cost solution that has a proven track recorvoip-info.org/setting-up-paging-with-a-sound-cardd. With a total investment of $80 for the phone, wire, and connectors you can have a basic paging system at your office. A second unit at a remote office or warehouse makes it easy to have paging across the street, or on the other side of the world.

• open up the phone and splice a connector jack in place of the builtin speaker. You can use a female RCA jack or a mini-stereo jack.
• jack can easily be mounted in the side of the case and used to connect to a traditional paging amplifier or amplified computer speakers.
• the reboot process as outlined on Asterisk phone grandstream budgetone works quite well for keeping these phones registered on the Asterisk. We’ve set them to reboot every four hours and have enjoyed over six months without a single user complaint.
• The Grandstream GXP-2000 would also work well for this- it has a 3.5mm audio jack built in. I have also read that the new redesigned BT100 series also has a headset jack.

The Grandstream GXP-2000 works very well for overhead paging. You can punch down on a 66 block a 3.5mm jack cable which then connects to your paging system. With the four sip accounts you can customize paging for different departments by having a different ring tone configured. I have this connected to an older Valcom 9970 Single Zone unit and two Handytone 488 attached to two 9 Zone Valcom 1109RTVAs. The 1109RTVA unit accepts your dtmf 0-9 (0 all call) to determine which zone to page. I can now page across the VPN to other buildings. Make sure you set the HT-488 FXO Port PSTN Silence Timeout to 10 seconds instead of 60 for paging. This reduces lockups. Also change FAX mode from t38 to Pass Thru. This is firmware 1.0.3.44 bootloader 1.0.8.11 – diver

We have worked with Grandstream to develop a dial plan example that lets you use both the built-in paging function as well as a dedicated prefix method for intercom Asterisk Intercom/Paging with Grandsteam (Revised for GXP-21xx Series Phones) 10-22-2010 – BEZ(zktech)

Paging as Ringer

In cases where paging and intercom are used over a Public Address Amplifier. The paging can also be utilized as ringer to let users know of incoming calls. This could be utilized in many ways from simply ringing, to actually automatically paging someone. Here is how to put it together in freepbx or trixbox:

1. Upload a ringing sound wav file as a System Recording
2. Upload a silence wav file as a System Recording
3. Go into the ring group settings and:
   1. 1. enable Confirm Calls
      2. append a ‘#’ to the end of the extension(s) for the PA-connected 299# for example
      3. for Remote Announce, select the “ringing” system recording
      4. for Too-Late Announce, select the “silence” system recording

See Also

• Asterisk cmd Page: Dialplan application that utilizes MeetMe to page one or more phones
• Asterisk phone door
• Cisco 7940-7960 auto-answer config
• Polycom auto-answer config
• Perl Program to implement paging via auto answer Polycom phones
• Asterisk Intercom/Paging with Grandsteam (Revised for GXP-21xx Series Phones) 10-22-2010 – BEZ(zktech)

Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones